Support for fax and modem in SIP/SIP-T networks and the interworking of these networks with ISUP+/BICC

ABSTRACT

The invention relates to a method for the transmission of call control parameters for switching over between a voice transfer mode and a data transfer mode between two media gateway controllers ( 6, 7 ), which are used by way of an IP network between two telecommunication terminal devices separated from any medium or bearer connection, whereby the call control parameters are transferred with conversion into the SIP or SIP_T protocol or from this into a standard signaling protocol.

CROSS REFERENCE TO RELATED APPLICATIONS

This application is the US National Stage of International ApplicationNo. PCT/DE03/03456, filed Oct. 17, 2003 and claims the benefit thereof.The International Application claims the benefits of German applicationNo. 10252989.2 DE filed Nov. 14, 2002, both of the applications areincorporated by reference herein in their entirety.

FIELD OF INVENTION

Support for fax and modem in SIP/SIP-T networks and the interworking ofthese networks with ISUP+/BICC.

BACKGROUND OF INVENTION

The invention relates to a method for the transmission of call controlparameters for switching over between a speech transmission mode and adata transmission mode between two media gateway controllers, which areused by way of an IP network between two telecommunication terminaldevices separated from any medium or bearer connection.

Service providers, in other words companies which, based for example onthe transport services of the partial networks (backbones) of theinternet, provide their customers or users with switched datacommunication facilities, are faced in the communications sectorprimarily with the challenges associated with introducing new fast andefficient services. In this situation, the network infrastructure forconnection network operators (carriers) is a significant cost factorwhereby cost savings can be achieved by means of successfuloptimization.

Before dealing with the question as to how a service provider can takeadvantage of the data signaling for service management solutions, it ishowever necessary to consider what the customer is expecting of aservice provider who is supplying value added facilities and services.At present, the expectation comprises a whole spectrum of integratedservices, a simple and uniform accounting process in real time, and alsocomprehensive access to measurable and recorded standards of service.

In order to nevertheless achieve an optimization of the operating costsin the case of a service provider despite their provision ofcomprehensive services, steps are now being taken to set upcommunication connections which perform the connection setup (call) andthe medium or bearer setups separately from one another. As a result, itis possible to revert to a relatively low-cost bearer technology forthrough-connection of the speech/data channel, whilst for example thesignaling is handled separately.

If a disconnection (decomposition) of the connection setup and themedium or bearer setup is carried out, a communication is then neededbetween at least two media gateway controllers which are used duringconnection setup for data signaling purposes.

The Diagram in FIG. 1 demonstrates how, in the case of such adisconnection of connection setup and bearer setup, the informationrequired for the setup of a communication connection between twotelecommunication terminal devices 1, 2 is exchanged between theindividual network components. In this situation, an A subscriberrequests a call setup to a telecommunication terminal 2 of a B user, bymeans of a telecommunication terminal 1 which is connected to a firstpublic telephone network (PSTN) 3 with an associated local switchingcentre (Local Exchange; LE) 5.

This call request results in a connection setup which takes place bymeans of two media gateway controllers (MGC) 6, 7. In this situation,information is transmitted by means of a corresponding signalingprotocol (Common Channel Signaling CCS:ISUP) to a first media gatewaycontroller 6. The latter communicates for its part with a second mediagateway controller 7 by way of a BICC CS2/ISUP+ interface via Q.765.5BAT (Bearer Application Transport). The second media gateway controller7 thus receives all the service or performance features which arerequired in order to setup the connection, and transmits thisinformation to a public telephone network 4 in which the terminal device2 of the B subscriber is located. The transmission takes place in turnby way of a corresponding signaling protocol.

On the basis of the prior art, it is known that by introducing suitablecommunication between a media gateway controller and the associatedmedia gateway the cost-effective bearer technologies such as IP can beapplied to through-connection of the speech/data channel. At the presenttime, there exists the ITU-T Standard Q.1902.X BICC CS2 (BearerIndependent Call Control Capability Set 2) with a special ServiceIndicator in the case of the MTP (Message Transfer Part), and Q.765.5BAT (Bearer Application Transport) which when employing IP-RTP as thebearer technology describes how its usual services are to be provided tothe end user in the telecommunication network when there is adisconnection between connection setups and through-connection of thespeech/voice channel. In the meantime, IETF has produced RFC 3204 (ISUPMIME Type) which enables the tunnel transport of ISUP messages in SIPmessages. Such SIP messages for the purpose of media gateway controllerintercommunication are also referred to as SIP-T messages.

A disadvantage that has become apparent in respect of the known solutionis the fact that with regard to intercommunication between media gatewaycontrollers themselves information can only be transmitted between themedia gateway controllers by way of the BICC CS2/ISUP+ interface viaQ.765.5 BAT (Bearer Application Transport). This holds true particularlywhen a modification of the bearer is required for the data transmission.Thus, in the case of a fax switchover for example, it is not possible atthe present time to transfer the information required for this purposebetween the media gateway controllers by means of a SIP protocol becausethe interworking of SIP to BICC is not yet defined in the ITU-TStandards for BICC CS2. The SIP protocol—in contrast to BICC—is suitablenot only as a subscriber protocol for voice connections but also formultimedia (voice and data) and will for this reason acquire anincreasing significance from the global viewpoint.

SUMMARY OF INVENTION

As a result of the described disadvantages arising from the prior art,the object of the present invention is to set down a method whereby itis possible to transmit messages between media gateway controllers byway of the SIP_T protocol and whereby it is also possible in particularin the case of a connection setup which is performed separately from thethrough-connection of a B-channel (bearer establishment) to transfer theinformation used for modification of the bearer via the BICC CS2/ISUP+interface of the media gateway controllers.

In this situation, messages are understood to comprise all the basicinformation that is transferred as call control parameters for examplebetween the media gateway controllers. In this situation, a delimitationshould be noted in particular from signals which are exchanged forexample for signaling purposes between the media gateway controllers.Such messages could for example contain the type of information whichindicates that a codec needs to be switched over on both sides for acommunication connection, as is required say in the case of a fax ormodem transmission.

The object is achieved by implementing the type of method described atthe beginning whereby the call control parameters are transmitted withconversion into the SIP or SIP_T protocol or from this into a standardsignaling protocol.

The advantage of the invention consists particularly in the fact that itcomprises a method which is especially easy to implement and equally aseffective and which allows a service provider to take advantage of thedata signaling for an extended range of service management solutions.The invention incorporates the fundamental concept of employing amapping of ISUP messages in the signaling protocol. The method proposesthe integration of the messages in the protocol for the management ofmultimedia sessions by inserting message cells containing the messagesto be transmitted into containers of the signaling protocol or onto theSDP level or BAT Q.765 level. As a result of the mapping, the messageswhich are required for modification of the bearer are inserted into theSDP level or BAT Q.765 level of the protocol for the management ofmultimedia sessions, which allows bearer-independent signalinginformation to be exchanged by way of the SIP, SIP-T or BICC CS2/ISUP+interface between the media gateway controllers.

Preferred developments of the invention are set down in the subclaims.

An especially preferred embodiment is thus provided in order to make useof IP technology for through-connection of speech/data channels in orderthat the data transfer takes place by way of an IP network afterestablishment of the communication connection between thetelecommunication terminal devices. The advantage of this embodimentconsists particularly in the fact that in this situation an especiallylow-cost and widespread bearer technology is used, which leads to a costsaving on the part of the service provider.

By particular preference, use is made for call control purposes of BICCprotocols (particularly the BICC CS2-ISUP+ protocol) as signalingprotocols, which are signaling protocols for operator-independent callcontrol. As a result, the network operator is in a position to offer acomplete set of PSTN/ISDN service including all supplementary servicesover a large number of transport networks. As a rule, however, othersignaling protocols can also be considered for use in this situation.

In a preferred embodiment of the present invention, provision is madewhereby, in the case of a fax transmission which is detected on one sideon the basis of a fax tone, this is signaled to the other side by way ofthe media gateway controller. In this situation, the message exchangedis the information indicating that the IP codec should be switched overon both sides of the connection, thereby enabling a fax transfer. Tothis end, the codec must be switched over to the G.711 or T.38 codec. Inaddition, the Maximum Jitter Buffer Size is increased on both sides andthe Silence Suppression is deactivated. The Echo Cancellation remainsunchanged. As a rule, the Echo Cancellation is activated for a voicecall, which also improves the quality of the data transmission betweenthe fax devices in the case of a fax call. The situation is similar fora modem transmission: When a modem tone is detected, the IP codec islikewise switched over, whereby only the codec G.711 can be used here.The Maximum Jitter Buffer Size is likewise increased and the SilenceSuppression is deactivated. The Echo Cancellation is also deactivated.

The task is now to transmit these call control parameters, whichhitherto have been present in the standard signaling protocol, in asuitable manner into the SIP or SIP_T protocol. This is done in the SDPpart of the SIP/SIP_T message in accordance with RFC2327. The proposedsolution functions for SIP and SIP_T, and the designation “SIP”therefore always stands for both interfaces in the following.

As an example of this the following table shows a possibleimplementation of the conversion of BICC parameters in the SDP part andvice versa, which is explained in detail afterwards. It should be notedthat the BICC parameters are shown in the left-hand column of the tableand the associated SDP part appears in the right-hand column. It shouldbe noted that when a “SIP:200 OK” acknowledgment message is converted inthe BICC, only one Action Indicator is set up in the SIP.

Q.765.5 Info Element APM SDP Element (re-INVITE) SIP (APP) BICC ActionIndicator The info element “Action Indicator” is implicitly transferredin the SDP session. Single Codec SDP Media Description Field,mediaformat (payload type number) SDP Attribute “rtpmap”(Codec Mapping)SDP Attribute “fmtp” (Codec parameter) Echo Cancellation SDP Attribute“ecan” (see RFC3108) Action Indicator successful 200 OK modified

According to the above example, the BICC CS2/ISUP+ message for a fax ormodem codec switchover contains the following info elements as perQ.765.5: (A) Action Indicator, (B) Single Codec and (C) EchoCancellation, whereby the latter is a proprietary element. These infoelements are discussed in detail in the following

(A) Action Indicator (Identifier 0x00000001):

The info element “Action Indicator” contains the value “modify codec”(0x00001011) if the codec switchover is initiated by a media gatewaycontroller. It is not converted to SDP but is regenerated on the otherside if a “SIP:re-INVITE” message is received with an SDP session whichcontains one of the codecs described below for the data transfer (seeSingle Codec). During the conversion of the info element “ActionIndicator” into the SIP protocol, the connection address of the RTP portremains unchanged.

The info element “Action Indicator” contains the value “successful codecmodification” (0x00001100) if the remote media gateway controller hassuccessfully performed the codec switchover and acknowledges the“SIP_T:re-INVITE” or SIP message. Here the initiating side receives a“SIP:200 OK” acknowledgment with an SDP session which contains preciselythe same codec for the data transfer and can thus regenerate the ActionIndicator. The connection address of the RTP port also remains unchangedin this case during the conversion of the info element “ActionIndicator” into the SIP protocol.

(B) Single Codec (Identifier 0x00000101):

The info element “Single Codec” is a proprietary identifier and containsthe codec which must be switched over to for the fax or modem datatransfer. In this situation, there are two possible codec values at thepresent time:

(i) “G.711 64 kbit/s A-law Modem Transparency”

This codec is transferred if the codec G.711 is to be used withoutSilence Suppression and with a high Maximum Jitter Buffer value for thefax or the modem transfer.

In a possible implementation of the method according to the invention, aproposal is made to convert this codec to SDP in accordance with theIETF Draft “draft-foster-mmusic-vbdformat-01.txt”. Accordingly, thiscould be done by:

Specifying a dynamic codec in the “m=” line:

m=audio 3456 RTP/AVP 99

Assigning the MIME subtype “vbd” (voiceband data traffic) to this codec:

a=rtpmap:99 vbd/8000

Assigning the “underlying” audio format (e.g. 8=PCMA) to this codec:

a=fmtp:99 8

The other side is thereby informed that the voice codec is to be set toG.711 for a data transmission. Thus the Maximum Jitter Buffer can alsobe automatically raised and the Silence Suppression can be deactivated.

(ii) “ITU-T Recommendation T.38 Codec Based on UDP”

It is sufficient here for example, in accordance with “ITURecommendation T.38, Amendment 2”, to resume a modified SDP MediaDescription with mediatype “image”, protocol “udptl” and payloadtype“t38”: m=image 3456 udptl t38

(C) Echo Cancellation (Identifier 0x11100001):

The info element “Echo Cancellation” contains the information thatdetermines whether the echo cancellation is to be activated ordeactivated on the other side. The info element “Echo Cancellation” isnot transmitted during fax switchover because the echo cancellation isnot modified in this case but remains as it was set for the voice call.With regard to a modem switchover (codec “G.711 64 kbit/s A-law modemtransparency”, see above), this element is transferred because the echocancellation must be deactivated in order to allow the data to betransmitted between the modems without errors.

In a possible implementation of the method according to the invention, aproposal is made to use the SDP attribute “ecan” from RFC3108(Conventions for the use of the Session Description Protocol for ATMBearer Connections) for the transmittal this information. In thissituation, the parameter “directionFlag” is set to “fb” (forward andbackward direction), the parameter “ecanEnable” is set according to theQ.765.5 info element to “on” or “off”, and the parameter “ecanType” isnot used and is set to “-”. Example: “a=ecan:fb on-”

With this proposed solution, it also becomes possible for messages to betransported by way of the SIP_T protocol between two media gatewaycontrollers which are required for the modification of the IP bearer fora fax or modem transfer. In addition, interworking between BICCCS2/ISUP+ and SIP-T is possible for these features. Furthermore, theproposed solution also enables interworking between an SIP terminaldevice and other IP-based solutions for these features.

BRIEF DESCRIPTION OF THE DRAWINGS

Preferred embodiments of the invention will be described in detail inthe following with reference to the drawings.

In the drawings:

FIG. 1 shows a diagram representing media gateway controllerintercommunication;

FIG. 2 shows a schematic representation of a first embodiment of theinvention; and

FIG. 3 shows a second embodiment of the invention.

DETAILED DESCRIPTION OF INVENTION

FIG. 1 shows a schematic representation of media gateway controllerintercommunication. This intercommunication is necessary in cases wherea disconnection (decomposition) of connection setup and medium or bearersetup is carried out. A subscriber terminal device 1 in a publictelephone network (PSTN) 3 requests a connection setup in order toestablish a communication connection by way of the associated localexchange (LE) 5 and transit exchange (TX) 9, which is performedseparately from the through-connection of a speech/data channel.

The connection setup takes place by means of signaling (CCS:ISUP) to theassociated media gateway controller (MGC) 6. This media gatewaycontroller 6 communicates with a second media gateway controller 7,whereby all signaling information is exchanged between the BICCCS2/ISUP+ interfaces of the two controllers 6, 7 by way of Q.765.5 BAT.The second media gateway controller 7 subsequently effects signaling forthe telecommunication terminal device 2 of the B subscriber.

FIG. 2 shows a schematic representation of a first embodiment of theinvention. With regard to the method represented, data is sent from thetelecommunication terminal device 1 of an A subscriber to a first mediagateway 10 where, on the basis of the data transferred, the ITU-Tstandard (or the coding type) of the data transmission is determined.The result is then transferred to a first media gateway controller 6,associated with the first media gateway 10, where a codec required forthe data transfer operation on the B-channel is determined on the basisof the ITU-T standard ascertained. The result of determining therequisite codec is transferred to a second media gateway controller 7 bymeans of corresponding signaling.

The second media gateway controller 7 controls an associated secondmedia gateway 11 which constitutes the interface to thetelecommunication terminal device 2 of a B subscriber. Subsequently, thefirst or second media gateway controller 6, 7 in the associated first orsecond media gateway 10, 11 respectively switches the codec for theB-channel over to the determined codec, required for the data transfer,by means of corresponding signaling.

On the basis of the ITU standard determined for the data transfer, thetransfer attributes required for optimum data transfer, particularlyMaximum Jitter Buffer Size, Silence Suppression and/or EchoCancellation, are also determined. This is also transferred with thesignaling from the first media gateway controller 6 to the second mediagateway controller 7.

The information transfer between the two media gateway controllers 6, 7occurs in the SDP part of the SIP-SIP-T message as per RFC2327. In thissituation, the conversion of the parameters into the SDP protocol takesplace in the manner previously described.

Following the transmission of the information between the two mediagateway controllers 6, 7, setting of the determined transfer attributesfor the codec for optimization of the data transfer operation takesplace by means of corresponding signaling from the media gatewaycontrollers 6, 7 to the associated media gateways 10, 11. The signalingfor conveying the result of the determination of the codec required forthe data transfer operation is thus effected in this situation by meansof a mapping of an ISUP message in a SIP message, particularly in an SDPpart of the SIP message as per RFC2327, by way of SIP interfaces of thefirst and second media gateway controllers 6, 7. The signaling betweenthe first and second media gateway controllers 6, 7 and the associatedfirst and second media gateways 10, 11 respectively is advantageouslyhandled by means of the MGC protocol/H.248. However, other protocols cannaturally also be considered for use.

FIG. 3 shows a schematic representation of a first embodiment of theinvention. In this embodiment, the subscribers' telecommunicationterminal device who is involved in the setup of a communicationconnection is a SIP client 12 which is connected for communicationpurposes by way of a SIP proxy 13 to the first media gateway controller6. It is accordingly assumed that the information required forconnection establishment is transferred from the SIP client 12 to thefirst media gateway controller 6 by means of a SIP protocol. With regardto the intercommunication between the two media gateway controllers 6,7, the BICC CS2/ISUP+ messages would be converted exactly as describedabove into SIP messages including the SDP session. This presupposes thatthe SIP client 12 supports fax or modem transmission.

Other embodiments of the method according to the invention are naturallyalso conceivable, however. Further possible interworking scenarios wouldbe SIP client intercommunication with a VoIP trunking subscriber,whereby the configuration should be organized as in the case of thesecond embodiment, except that in this situation a single media gatewaycontroller would suffice instead of two media gateway controllersbecause one BICC CS2/ISUP+ is dispensed with between exchange signaling.

SIP client intercommunication with an access gateway (HIA7600) or withan H.323 subscriber, VoDSL subscriber or SIP client would also beconceivable.

1. A method for transmitting call control parameters between two mediagateway controllers, wherein the call control parameters enable aswitchover between a voice transfer mode and a data transfer modeseparate from a setup of a medium or bearer connection comprising:providing a first media gateway controller and a second media gatewaycontroller, the first and second controllers included in an internetprotocol (IP) network; providing a terminal device; converting the callcontrol parameters into a session initiation protocol (SIP) or sessioninitiation protocol for telephones (SIP-T) message, the call controlparameters including a parameter for indicating a codec switchover ofthe bearer connection in the form of an action indicator in accordancewith Q.765.5; transmitting a session message containing the convertedcall control parameters; converting the session message containing theconverted call control parameters into a message based on a standardsignaling protocol to be transmitted between the two media gatewaycontrollers; and transmitting the message between the two media gatewaycontrollers based on the standard protocol with the call controlparameters.
 2. The method according to claim 1, wherein the standardsignaling protocol is selected from the group consisting of bearerindependent call control protocol and ISDN user part (ISUP).
 3. Themethod according to claim 1, wherein the call control parameters map toa session description protocol part of the session message in accordancewith Internet Engineering Task Force Request for Comment No. 2327(RFC-2327), the session message being transmitted between thecontrollers.
 4. The method according to claim 1, wherein the callcontrol parameters include a switchover parameter indicating a codecswitchover of the bearer connection, the switchover parameter implicitlyrepresented in a session description protocol part of the sessionmessage and explicitly represented as an action indicator in thestandard signaling protocol.
 5. The method according to claim 4, whereinthe first media gateway controller initiates the codec switchover andsends a session internet protocol reinvite (SIP:re-INVITE) message alongwith the session description protocol part, the session descriptionprotocol part indicating a data transfer codec.
 6. The method accordingto claim 5, wherein the second media gateway controller receives theSIP:re-INVITE message along with the session description protocol part,the session description protocol (SDP) part indicating the data transfercodec, the second media gateway controller generating a first value inthe standard signaling protocol.
 7. The method according to claim 4,wherein the second media gateway controller performs the codecswitchover and sends a SIP:re-INVITE acknowledgement message with thesession description protocol part indicating a data transfer codec. 8.The method according to claim 7, wherein the first media gatewaycontroller receives a SIP:re-INVITE acknowledgement message along withthe session description protocol part, the session description protocolpart indicating the data transfer codec, whereby the first media gatewaycontroller generates a second value in the standard signaling protocol.9. The method according to 3, wherein the call control parametersinclude a codec parameter that provides a codec for the data transfermode.
 10. The method according to claim 9, wherein the codec parameteris transmitted within the session description protocol part of thesession message, the session description protocol part including a mediadescription field, a codec mapping session description protocolattribute (rtpmap), and a session description protocol attributeparameter (fmtp).
 11. The method according to claim 9, wherein the codecparameter is transmitted in the standard signaling protocol, andcontains a single codec parameter.
 12. The method according to claim 3,wherein the call control parameters contain an echo cancellationparameter indicating activation or deactivation of an echo cancellation.13. The method according to claim 12, wherein the echo cancellationparameter indicates activation during a switchover to a fax transfermode, and the echo cancellation parameter indicates deactivation duringa switchover to a modem transfer mode.
 14. The method according to claim13, wherein the echo cancellation parameter is transmitted in thesession description protocol part of the session message that includesan echo cancellation session description protocol attribute (ecan). 15.The method according to claim 13, wherein the call control parameterscontain an echo cancellation transmitted in the standard signalingprotocol.
 16. The method according to claim 9, wherein the switchover tothe data transfer mode causes a bearer connection codec change to G.711,a maximum jitter buffer size of the bearer connection is increased, anda silence suppression of the bearer connection is deactivated.
 17. Themethod according to claim 9, wherein the switchover to the data transfermode is a T.38 codec.
 18. The method according to claim 3, wherein acodec is communicated from the controllers to an associated mediagateway in the standard signaling protocol selected from the groupconsisting of media gateway control protocol (MGCP) and H.248, the codecis communicated after the transfer of the call control parametersbetween the media gateway controllers.
 19. A communications networkcomprising: a plurality of telecommunication terminals connected to apublic switched telephone network; a media gateway controllercommunicating with the public switched telephone network via a standardsignaling protocol selected from the group consisting of bearerindependent call control protocol and ISUP; a plurality of mediagateways communicating with each other over an internet protocol networkand controlled by one or more media gateway controllers; the mediagateway controller transferring call control parameters, the callcontrol parameters being mapped into a session description protocol partof a session message in accordance with RFC-2327 and including aparameter for indicating a codec switchover of the bearer connection inthe form of an action indicator in accordance with Q.765.5; and thesession message being transmitted between the media gateway controllersaccording to the session description protocol.
 20. The network accordingto claim 19, wherein the call control parameters include a codecparameter that provides a codec for the data transfer mode, the codecparameter is transmitted within the session description protocol part ofthe session message, the session description protocol part including amedia description field, a session description protocol attribute ofrtpmap, and a session description protocol attribute of fmtp.
 21. Acommunications network comprising a first media gateway controller and asecond media gateway controller, converting call control parameters intoSIP or SIP-T message, the call control parameters including a parameterfor indicating a codec switchover of the bearer connection in the formof an action indicator in accordance with Q.765.5; transmitting the SIPor SIP-T message containing the converted call control parameters;converting the SIP or SIP-T message containing the converted callcontrol parameters into a message based on a standard signaling protocolto be transmitted between the two media gateway controllers; andtransmitting the message between the two media gateway controllers basedon the standard protocol with the call control parameters, wherein thecall control parameters enable a switchover between a voice transfermode and a data transfer mode separate from a setup of a medium orbearer connection.